Audio Data Packet Format and Decoding Method thereof and Method for Correcting Mobile Communication Terminal Codec Setup Error and Mobile Communication Terminal Performance Same

ABSTRACT

Disclosed is an audio data packet format for transmitting an IYIPEG-4 HE-AAC frame via a voice channel of a mobile communication network, a method for decoding the audio data packet format, a method for correcting a codec setup error by identifying a codec used to encode sound source data inserted into a data field of voice slot data, based on the sequence number of the voice slot data, and correcting the codec setup error when a codec set up in a mobile communication terminal is different from the codec used to encode the sound source data, and a mobile communication terminal adapted to correct a codec setup error.

TECHNICAL FIELD

The present invention relates to an audio data packet format, a methodfor decoding the audio data packet format, a method for correcting acodec setup error, and a mobile communication terminal adapted tocorrect the codec setup error. More particularly, the present inventionrelates to an audio data packet format for transmitting an MPEG-4 HE-AAC(High Efficiency Advanced Audio Coding) frame via a voice channel of amobile communication network and a method for decoding the same. Inaddition, the present invention relates to a method for correcting acodec setup error by identifying a codec used to encode sound sourcedata inserted into a data field of voice slot data, based on thesequence number of the voice slot data, and correcting the codec setuperror when a codec set up in a mobile communication terminal isdifferent from the codec used to encode the sound source data, as wellas a mobile communication terminal adapted to correct a codec setuperror in the same method.

BACKGROUND ART

As generally known in the art, various services are provided via mobilecommunication networks in line with rapid development of technologiesassociated with computers, electronics, and communications. The mostbasic type of mobile communication service is a voice communicationservice, which enables users to communicate via mobile communicationterminals regardless of time and place. In addition, a text messageservice complements the voice communication service. A wireless Internetservice has recently been provided and enabled users of a mobilecommunication terminal to access the Internet via mobile communicationnetworks.

As a result, subscribers to a mobile communication service can not onlycommunicate with desired partners regardless of time and place, but alsoreceive various types of daily information (e.g. news, weather, sports,stocks, exchange rate, traffic) in the form of texts, voices or imagesvia wireless Internet access.

Due to the recent development of communication technology, mobilecommunication services are shifting from voice communication services tomultimedia communication services for transmitting circuit data orpacket data, for example. Recently, the IS-95C network has evolved fromconventional IS-95A and IS-95B networks and is capable of providingwireless Internet services at a data transmission rate of up to 307.2Kbps, which is much faster than that supported by the conventionalnetworks (14.1 Kbps and 56 Kbps). Particularly, IMT-2000 services cannot only improve the quality of conventional voice communication and WAPservices, but also provide various multimedia services (e.g. audio ondemand, video on demand) at a higher rate.

Recently, ringback tone or color ringback tone services are drawing muchattention. These services provide originators with various soundsources, which have been selected by recipients, as a ringback tone viaa communication terminal. Particularly, conventional ringback tones,which are mechanical and monotonous, are replaced with various types ofmusic or sound (e.g. popular music, sound of nature), which have beenrecorded by recipients. Upon hearing these novel ringback tones,originators get special impressions.

However, current ringback tone services have a problem in that outputtedsound sources do not have excellent quality and, if a frame is lostwhile sound source data is transmitted for providing a ringback tone,mobile communication terminals cannot recognize the loss. This resultsin erroneous decoding.

DISCLOSURE OF THE INVENTION

Accordingly, the present invention has been made to solve theabove-mentioned problems occurring in the prior art, and an object ofthe present invention is to provide an audio data packet format fortransmitting audio data (e.g. ringback tone) via a voice channel byusing an MPEG-4 HE-AAC audio codec frame so as to improve the quality ofaudio data.

Another object of the present invention is to provide a method fordecoding an audio data packet quickly and accurately by adding asequence number field when an MPEG-4 HE-AAC audio codec frame istransmitted after being divided into voice slot data.

Still another object of the present invention is to provide a method forcorrecting a codec setup error by identifying a codec used to encodesound source data inserted into a data field of voice slot data, basedon the sequence number of the voice slot data, and correcting the codecsetup error when a codec set up in a mobile communication terminal isdifferent from the codec used to encode the sound source data, as wellas a mobile communication terminal adapted to correct a codec setuperror in the same method.

In order to accomplish this object, there is provided an audio datapacket format comprising a first field for containing an audio data of apredetermined size to be transmitted, the audio data being fragmentedfrom an audio frame; and a second field for specifying an order ofdivided audio data, wherein the audio frame is based on an MPEG-4 HE-AAC(High Efficiency Advanced Audio Coding) scheme.

In accordance with another aspect of the present invention, there isprovided a method for decoding an audio data packet having a first fieldfor containing an audio data of a predetermined size, the audio databeing fragmented from an audio frame and a second field for specifyingan order of divided audio data, the method including the steps of: (a)initializing a reference sequence number by using a decoder of a mobilecommunication terminal; (b) checking a second field of received audiodata to confirm whether or not a first bit stream of the audio frame istransmitted; (c) decoding data preceding currently received data when afirst bit stream of a new audio frame is transmitted; (d) storing thecurrently received data; (e) resetting the reference sequence number;and (f) returning to step (b).

In accordance with another aspect of the present invention, there isprovided a method for correcting a codec setup error in a mobilecommunication terminal by analyzing voice slot data received from amobile communication network while a multimedia audio codec is set up inthe mobile communication terminal and identifying a codec used to encodesound source data inserted into a data field of the voice slot data toconfirm whether or not the mobile communication terminal and the mobilecommunication network have an identical codec, the method comprising thesteps of: (a) checking a sequence number inserted into the data field ofevery received voice slot data and determining that there is a codecsetup error when the voice slot data is not received in order; and (b)replacing the multimedia audio codec with a voice codec, when it hasbeen determined that there is a codec setup error.

In accordance with another aspect of the present invention, there isprovided a method for correcting a codec setup error in a mobilecommunication terminal by analyzing voice slot data received from amobile communication network while a voice codec is set up in the mobilecommunication terminal and identifying a codec used to encode soundsource data inserted into a data field of the voice slot data so as toconfirm whether or not the mobile communication terminal and the mobilecommunication network have an identical codec, the method comprising thesteps of: (a) checking a sequence number SEQ inserted into the datafield of every received voice slot data and determining that there is acodec setup error when the voice slot data is received in order; and (b)replacing the voice codec with a multimedia audio codec, when it isdetermined that there is a codec setup error.

In accordance with another aspect of the present invention, there isprovided a mobile communication terminal for correcting a codec setuperror by analyzing voice slot data received from a mobile communicationnetwork and identifying a codec used to encode sound source datainserted into a data field of the voice slot data to confirm whether ornot the mobile communication terminal and the mobile communicationnetwork have an identical codec, the mobile communication terminalcomprising: a voice codec for decoding and outputting voice data beinginserted into the voice slot data and being transmitted during voicecommunication; a multimedia audio codec for decoding and outputtingmusic data transmitted by a CRBT (Color Ring Back Tone) server providinga CRBT service, the music data being inserted into the voice slot dataand being transmitted; and a codec control unit for performing a firstfunction of driving the voice codec when a control message informing thestart of voice communication is received from the mobile communicationnetwork, driving the multimedia audio codec when a control messageinforming the transmission of a ringback tone is received, checking asequence number inserted into the data field of every received voiceslot data when the voice slot data is received while the multimediaaudio codec is in operation, determining that there is a codec setuperror when the voice slot data is not received in order, restricting theoperation of the multimedia audio codec, and letting the voice codec inoperation, the codec control unit performing a second function ofchecking a sequence number inserted into the data field of everyreceived voice slot data when the voice slot data is received while thevoice codec is in operation, determining that there is a codec setuperror when the voice slot data is received in order, restricting theoperation of the voice codec, and driving the multimedia audio codec.

BRIEF DESCRIPTION OF THE DRAWINGS

The above and other objects, features and advantages of the presentinvention will be more apparent from the following detailed descriptiontaken in conjunction with the accompanying drawings, in which:

FIG. 1 shows a construction of a conventional EVRC data packet;

FIG. 2 shows the construction of an audio data packet format accordingto the present invention;

FIG. 3 shows an example of an audio data packet format including a frameboundary section identifier;

FIG. 4 is a flowchart showing a method for decoding an audio data packetaccording to the present invention;

FIG. 5 is a flowchart showing a method for correcting an codec setuperror in a mobile communication terminal providing a ringback toneservice according to a preferred embodiment of the present invention,wherein, when an HE-AAC codec has been set up in the mobilecommunication terminal, a sequence number field is used to check whetheror not the mobile communication system and the mobile communicationterminal have the same codec;

FIG. 6 is a flowchart showing a method for correcting an codec setuperror in a mobile communication terminal providing a ringback toneservice according to a preferred embodiment of the present invention,wherein, when an EVRC has been set up in the mobile communicationterminal, a sequence number field is used to check whether or not themobile communication system and the mobile communication terminal havethe same codec; and

FIG. 7 briefly shows the construction of a mobile communication terminaladapted to correct a codec setup error while providing a ringback toneservice according to a preferred embodiment of the present invention.

BEST MODE FOR CARRYING OUT THE INVENTION

Hereinafter, a preferred embodiment of the present invention will bedescribed with reference to the accompanying drawings. In the followingdescription and drawings, the same reference numerals are used todesignate the same or similar components, and so repetition of thedescription on the same or similar components will be omitted.

FIG. 1 shows a construction of a conventional EVRC data packet.

Most mobile communication terminals employ a voice codec with variabletransmission rates, such as an EVRC (Enhanced Variable Rate Codec), inorder to provide efficient voice communication via optimized wirelesschannels. The EVRC digitally converts voices used by digital mobilecommunication systems and has a transmission rate of 8 Kbps. The EVRCcan efficiently keep sound quality from degrading. In addition, the EVRCvariably encodes voice information depending on the amount ofinformation. Particularly, the EVRC encodes voice at a low rate whenthere is a smaller volume of information (i.e. while the talker issilent) and at a high rate when there is a large amount of information.As such, the EVRC has a better voice encoding efficiency than when theencoding rate is constant. This expands the capacity of mobilecommunication systems and reduces the power consumption.

FIG. 1 shows the construction of a data packet used for the EVRC.Although an actual EVRC data packet includes two bypass frames of 80bytes, only one frame is shown in FIG. 1 for clarity.

A bypass frame includes a preamble field of 32 bits, a message headerfield of 32 bits, an encoding packet data field of 352 bits, a dummyfield of 194 bits, and a CRC (Cyclic Redundancy Check) field of 30 bits.The encoding packet data field includes a header field of 8 bytes, avoice packet field of 34 bytes, and a dummy field of 2 bytes. The voicepacket field includes a payload field of 171 bits and a dummy field of101 bits so that compressed sound source data can be loaded on thepayload field.

As such, the portion of an EVRC data packet occupied by actual datacorresponds to 171 bits. According to the present invention, an HE-AACframe for audio data transmission is included in that portion andtransmitted, in order to improve the quality of audio data (e.g.ringback tone).

As used herein, AAC (Advanced Audio Coding) refers to a coding schemefor digital audio signals and has been declared an internationalstandard based on MPEG of ISO/IEC. AAC frames have a variable sizedepending on the compression rate. This means that the entire filecapacity can be reduced substantially. Compared with MP3 files, the filecapacity of AAC frames can be reduced to 30%. In addition, the AACscheme applies TNS (Temporal Noise Shaping) and prediction techniques soas to improve the sound quality. The TNS is one of quantizationcorrection techniques and can intelligently reduce errors occurring whencontinuous analog data is converted into digital data of 0 and 1 so asto reduce noise and reproduce near original sound. The predictiontechnique stores numeric values corrected by the TNS. Particularly,previously corrected information is stored and used later when the sametype of data appears. When a sound comes to have a different correctionvalue during a quantization process, it may be regarded a differentsound. The prediction technique avoids this. As such, the AAC hasexcellent sound quality over MP3.

Due to such excellent performance and high quality, the AAC has beenadopted by MPEG-4, 3GPP, and 3GPP2 standards and is drawing muchattention as a new type of audio codec for Internet, wireless, anddigital broadcasting divisions. Furthermore, an MPEG4 HE-AAC(hereinafter, referred to as HE-AAC) has evolved from the AAC and iscapable of providing sound quality of CD-grade even at a lowtransmission rate. It is expected that, if the HE-AAC codec is appliedto a ringback tone service, for example, it can guarantee very highsound quality.

In order to apply the HE-AAC codec to a ringback tone service, forexample, the data transmission rate necessary for real-time decodingshould conform to the transmission rate of CDMA voice communicationchannels. In addition, HE-AAC frames should be divided into bit streamsand transmitted according to the slit structure of the CDMA voicecommunication channels. Particularly, information of up to 171 bitsshould be transmitted for 20 ms in conformity with the transmission rateof 8 Kbps and the channel slot structure as required by the CDMA voicecommunication channels.

FIG. 2 shows the construction of an audio data packet format accordingto the present invention.

When an HE-AAC frame is to be transmitted in an EVRC data packet formatused for CDMA voice communication channels, the frame should be includedin a payload field as shown in FIG. 1. The HE-AAC frame has a variablelength and is divided into such a size that it can be transmitted via a20 ms slot. Then, the frame is transmitted as bit streams via a wirelesslink. After being transmitted to a mobile communication terminal, theHE-AAC bit streams are combined by the terminal and reconstructed as theHE-AAC frame.

During transmission of the HE-AAC bit streams, a base station controlleradds multiplexing rate mode information (field M) of 1 bit, CRCinformation of 12 bits, and encoder tail information (field T) of 8 bitsto each bit stream.

In order to efficiently map an HE-AAC frame onto a CDMA voice channelslot, it should be processed by byte. For this mapping, 168 bits (21bytes) of a data field (171 bits) is used, and the remaining 3-bit dataarea remains as a free space, which may be used for another purpose.According to the present invention, the free space is used as a sequencenumber field SEQ.

It is impossible that a single CDMA voice slot includes an entire HE-AACframe. Therefore, the HE-AAC frame should be divided into at least onebit stream and transmitted. During this process, two HE-AAC frames mayappear consecutively within the same CDMA voice slot. The consecutiveHE-AAC frames are differentiated from each other by a decoder of themobile communication terminal, which searches for an ADTS (Audio DataTransport Stream) as an identifier for differentiating frames.

The ADTS header is basically included in the HE-AAC frame. The ADTSheader includes a Syncword field of 12 bits; an ID field of 1 bit; aLayer field of 2 bits; a Protection_Absent field of 1 bit; aProfile_ObjectType field of 2 bits; a Private_bit field of 1 bit; aSampling_Frequency_Index field of 4 bits; a Channel_Configuration fieldof 3 bits; a Home field of 1 bits; and an Original/Copy field of 1 bit.

The decoder of the mobile communication terminal has no information onwhich part of received data the ADTS header is located. Therefore, thedecoder checks the frame differentiation identifier of every receiveddata. This increases the processing load. Even when the wireless channelenvironments get worse and a transmission packet is lost, the modem ofthe mobile communication terminal does not transmit error data to thedecoder, which is still unaware of the loss. As a result, the length orcontent of the frame may be varied during an HE-AAC decoding process.

Therefore, in order to correct the decoding error resulting from thewireless channel error, to differentiate the HE-AAC frames efficiently,and to specify the order of a plurality of bit streams, into which asingle HE-AAC frame has been divided, 3 bits (free bits) of the payload(171 bits) are designated as a sequence number field SEQ. This solvesthe problems of increased load due to frame differentiation, as well asthe decoding error.

For example, the sequence number field SEQ may contain binary digitsfrom ‘000’ to ‘111’. Information of ‘000’ is inserted into a slot, whichcontains a newly beginning HE-AAC frame. The next voice slot of the sameHE-AAC frame is given a value increased by 1. When the next HE-AAC frameis inserted into the voice slot, it is designated as ‘000’ again.

Based on the sequence number, the decoder of the mobile communicationterminal can easily determine whether or not a voice slot, whichincludes an HE-AAC frame, has a loss and easily obtain informationregarding the beginning of the next HE-AAC, i.e. which part of the voiceslot data includes the ADTS header. Since the location of the SEQ fieldis fixed, the decoder searches received data and, if the SEQ field hasbeen designated as ‘000’, determines that the ADTS header is at thefront end of the corresponding bit stream.

In summary, the present invention employs an EVRC format so as totransmit audio data packets. The first field of the entire areadesignated as the payload field is used to transmit data, and the secondfield is used to transmit the sequence number. Upon receiving an audiodata packet, the mobile communication terminal can easily determinewhether or not the HE-AAC frame has a loss, clearly differentiate framesfrom each other, and quickly reconstruct the HE-AAC frame. For example,168 bits are allocated to the first field, and 3 bits to the secondfield.

FIG. 3 shows an example of an audio data packet format including a frameboundary section identifier.

The ADTS header, which is an identifier for differentiating HE-AACframes, is located at the front end of the first bit stream, i.e. bitstream having a SEQ field designated as ‘000.’ As such, the SEQ field isused to differentiate frames so that, if a slot including an ADTS headeris lost, that slot can be located by simply checking the sequence numberfield without having to search the data stream within all voice slotsfollowing the lost slot.

FIG. 4 is a flowchart showing a method for decoding an audio data packetaccording to the present invention.

The decoder of a mobile communication terminal initializes a referencesequence number value for differentiating frames or specifying the orderof bit streams (S101).

Upon receiving audio data from a mobile communication network (S103),the decoder checks the sequence number field SEQ of the received dataand confirms whether or not the sequence number is 0 or 000 (binarydigit), which means that a new frame has been received (S105).

If it has been confirmed that the sequence number is 0, the decoderchecks the ADTS header (S107) and stores all data preceding the ADTSheader information in the buffer (S109). Then, the decoder performsdecoding by coupling voice data stored in the buffer (S111).

The decoder stores remaining information, including the ADTS header, inthe buffer (S113) and resets the reference sequence number so that, whenthe next HE-AAC frame is received, its sequence number can be comparedwith the reference sequence number (S115). In this case, the referencesequence number is modified into the sequence number included in thecurrently received data.

After resetting the reference sequence number, the decoder stands byuntil the next data is received (S103) Then, the subsequent steps arerepeated.

If is has been confirmed in step S105 that the sequence number of thereceived data is not 0, the decoder checks whether or not the sequencenumber of the currently received bit stream is equal to the referencesequence number plus 1 (S117). In that case, i.e. if it has beenconfirmed that the currently received bit stream belongs to the sameframe as the previously received bit stream and directly succeeds it,the decoder stores the received data (S119). The decoder modifies thereference sequence number into the sequence number of the currentlyreceived bit stream (S121) and returns to step S103.

If it has been confirmed in step S117 that the sequence number of thecurrently received bit stream is not equal to the reference sequencenumber plus 1, the decoder determines that an error has occurred in datatransmission and deletes the received data (S123). Then, the decoderresets the reference sequence number (S125). Particularly, the referencesequence number is modified into the highest value that the sequencenumber field can indicate so that, since an error has occurred, decodingdoes not begin until a new HE-AAC frame is received.

Although not shown in the drawings, according to a preferred embodimentof the present invention, it is also possible to receive data, compareits sequence number with the reference sequence number, and notify theexchanger of a missing sequence number, if any, so that the lost datacan be retransmitted.

The above-mentioned method is advantageous in that it is possible toprevent the length or content of frames from varying during a decodingprocess due to the loss of HE-AAC frames. In addition, the amount ofprocessing required of the mobile communication terminal can be reducedby locating a CDMA voice slot including ADTS header information.

However, application of the method to a ringback tone service may causethe following problems.

When a ringback tone service is provided, a multimedia audio codec (e.g.HE-AAC) is used in a ringback tone section, and a voice codec (e.g.EVRC) is used in a speech section. If the codec setup of the mobilecommunication terminal is different from that of the system, the servicecannot be successful however efficient this scheme may be.

Therefore, a method for correcting a codec setup error in a mobilecommunication terminal using a sequence number field will now bedescribed.

FIG. 5 is a flowchart showing a method for correcting an codec setuperror in a mobile communication terminal providing a ringback toneservice according to a preferred embodiment of the present invention,wherein, when an HE-AAC codec has been set up in the mobilecommunication terminal, a sequence number field is used to check whetheror not the mobile communication system and the mobile communicationterminal have the same codec.

As mentioned with reference to FIGS. 2 and 3, when data distributedwithin and mapped onto a CDMA voice slot is an HE-AAC frame encoded byan HE-AAC codec, each CDMA voice slot data transmitted from the mobilecommunication network to the mobile communication terminal has asequence number assigned thereto in an order.

Therefore, the mobile communication terminal searches for the sequencenumber field of the CDMA voice slot data and, if the CDMA voice slotdata has not been received in the right order, determines that an errorhas occurred in the setup of the codec, which is then replaced with avoice codec (i.e. EVRC codec).

More particularly, upon receiving CDMA voice slot data from the basestation (S500), the mobile communication terminal designates thesequence number SEQ of a sequence number field, which has been insertedinto a data field of the received CDMA voice slot data, as a referencesequence number Bseq and initializes a codec counter Ecount (S502).

Upon receiving the next CDMA voice slot data (S504), the terminaldetermines whether or not the sequence number SEQ of the received CDMAvoice slot data is larger than the reference sequence number Bseq by 1(S506). If not, the terminal determines whether or not the sequencenumber SEQ is ‘000’ (S508).

If the sequence number SEQ is not ‘000’, the terminal increases thecodec counter by 1 (S510) and determines whether or not the codeccounter is equal to or larger than a predetermined number N (S512).

If it has been determined that the codec counter is equal to or largerthan the predetermined number N, the terminal confirms that there is acodec setup error. Then, the terminal replaces the currently set HE-AACcodec with a voice codec, i.e. EVRC codec (S516).

If it has been determined in step S506 that the sequence number SEQ islarger than the reference sequence number Bseq by 1, if it has beendetermined in step S508 that the sequence number SEQ is ‘000’, or if ithas been determined in step S512 that the codec counter is smaller thanthe predetermined number N, the terminal designates the receivedsequence number SEQ as the reference sequence number Bseq (S514) andproceeds to step S504.

The predetermined number N is selected so as to differentiate a codecsetup error from a loss of CDMA voice slot data, and is preferably equalto or larger than 2.

FIG. 6 is a flowchart showing a method for correcting an codec setuperror in a mobile communication terminal providing a ringback toneservice according to a preferred embodiment of the present invention,wherein, when an EVRC has been set up in the mobile communicationterminal, a sequence number field is used to check whether or not themobile communication system and the mobile communication terminal havethe same codec.

Upon receiving CDMA voice slot data at least a predetermined number oftimes in an order, the mobile communication terminal confirms that thecodec of the mobile communication system is an HE-AAC codec and replacesthe currently set EVRC codec with the HE-AAC codec.

More particularly, upon receiving CDMA voice slot data from the basestation (S600), the mobile communication terminal designates thesequence number SEQ of a sequence number field, which has been insertedinto a data field of the received CDMA voice slot data, as a referencesequence number Bseq and initializes a codec counter Ecount (S602).

Upon receiving the next CDMA voice slot data (S604), the terminaldetermines whether or not the sequence number SEQ of the received CDMAvoice slot data is larger than the reference sequence number Bseq by 1(S606). If so, the terminal increases the codec counter Ecount by 1(S608) and determines whether or not the codec counter Ecount is equalto or larger than a predetermined number N (S612).

If it has been determined in step S606 that the sequence number SEQ isnot larger than the reference sequence number Bseq by 1, the terminaldetermines whether or not the sequence number SEQ is ‘000’ (S610) and,if so, proceeds to step S612.

If it has been determined in step S610 that the sequence number SEQ isnot ‘000’, or if it has been determined in step 612 that the codeccounter Ecount is smaller than the predetermined number N, the terminaldesignates the received sequence number SEQ as the reference counter andproceeds to step S604 (S614).

If it has been determined in step S612 that the codec counter Ecounteris equal to or larger than the predetermined number N, the terminalconfirms that the codec of the mobile communication system is an HE-ACCcodec and replaces the EVRC voice codec, which has been currently set upin the mobile communication terminal, with the HE-AAC codec (S616).

FIG. 7 briefly shows the construction of a mobile communication terminaladapted to correct a codec setup error while providing a ringback toneservice according to a preferred embodiment of the present invention.

The mobile communication terminal adapted to correct a codec setup errorwhile providing a ringback tone service according to a preferredembodiment of the present invention includes an RF signal processingunit 700, a channel decoder 710, a signal processing unit 720, a codeccontrol unit 730, a voice codec 740, and a multimedia audio codec 750.

The RF signal processing unit 700 is adapted to convert wirelesssignals, which have been received via an antenna, into digital signals.The channel decoder 710 is adapted to decode digital signals, which havebeen received from the RF signal processing unit 700, so that they aredivided into signal and data portions. The signal portion is transmittedto the signal processing unit, and the data portion to the codec controlunit 730.

Upon receiving a control message (alerting info message) informing thatvoice communication begins from the mobile communication network, thecodec control unit 730 drives the voice codec 750. Upon receiving acontrol message informing that a ringback tone is transmitted (databurst message informing of setup of a multimedia audio codec), the codeccontrol unit 730 drives the multimedia audio codec 740.

Particularly, if the codec control unit 730 receives voice slot datawhile the multimedia audio codec 740 is driven, it checks the sequencenumber inserted into the data field of every received voice slot data.If the voice slot data has not been received in the right order, thecodec control unit 730 determines that there is a codec setup error.Then, the codec control unit 730 limits the driving of the audio codec740 and drives the voice codec 750 (first function) in the same methodas described with reference to FIG. 4.

If the codec control unit 730 receives voice slot data while the voicecodec 750 is driven, it checks the sequence number inserted into thedata field of every received voice slot data. If the voice slot data hasbeen received in the right order, the codec control unit 730 determinesthat there is a codec setup error. Then, the codec control unit 730limits the driving of the voice codec 750 and drives the multimediaaudio codec 740 (second function) in the same manner as described withreference to FIG. 5.

The voice codec 740 is adapted to decode voice data, which istransmitted while being inserted into voice slot data during voicetransmission. An EVRC codec is preferably used as the voice codec 740.

The multimedia audio codec 750 is adapted to decode music datatransmitted by a CRBT server, which provides a CRBT service. Althoughany type of audio codec may be used as the multimedia audio codec 750,an HE-AAC codec is preferably used.

Although a preferred embodiment of the present invention has beendescribed for illustrative purposes, those skilled in the art willappreciate that various modifications, additions and substitutions arepossible, without departing from the scope and spirit of the inventionas disclosed in the accompanying claims.

INDUSTRIAL APPLICABILITY

As mentioned above, the present invention is advantageous in that, byusing an HE-AAC mode for an audio service (e.g. ringback tone, colorringback tone), the quality of audio signals is improved remarkably. Inaddition, the use of an EVRC data format and the application of asequence number field when an HE-AAC frame is transmitted reduce thetime necessary to decode the HE-AAC frame. Furthermore, even when datahas been lost, it can be spotted quickly. This reduces the service timeand increases the rate.

The improved boundary search for differentiation between HE-AAC framesand the higher stability of the decoder guarantee an efficient ringbacktone service.

In addition, even when the codec used by the mobile communicationnetwork differs from that used by the mobile communication terminal dueto an error occurring while a call is processed in the case of aringback tone service using a multimedia audio codec for a ringback tonesection and a voice codec for a voice communication section, the mobilecommunication terminal can make a proper decision by itself and modifythe codec setup so that the ringback tone service can be providedwithout interruption.

1. An audio data packet format comprising: a first field for containingan audio data of a predetermined size to be transmitted, the audio databeing fragmented from an audio frame; and a second field for specifyingan order of divided audio data, wherein the audio frame is based on anMPEG-4 HE-AAC (High Efficiency Advanced Audio Coding) scheme.
 2. Theaudio data packet format as claimed in claim 1, wherein the audio frameis contained in a payload field of an EVRC (Enhanced Variable RateCodec) data packet.
 3. The audio data packet format as claimed in claim1, wherein the audio frame is divided by byte and transmitted via thefirst field.
 4. The audio data packet format as claimed in claim 1,wherein the first field includes an ADTS (Audio Data Transport Stream)header when a value allocated to the second field indicates a firstdivided data of the audio frame.
 5. A method for decoding an audio datapacket having a first field for containing an audio data of apredetermined size, the audio data being fragmented from an audio frameand a second field for specifying an order of divided audio data, themethod comprising the steps of: (a) initializing a reference sequencenumber by using a decoder of a mobile communication terminal; (b)checking a second field of received audio data to confirm whether or nota first bit stream of the audio frame is transmitted; (c) decoding datapreceding currently received data when a first bit stream of a new audioframe is transmitted; (d) storing the currently received data; (e)resetting the reference sequence number; and (f) returning to step (b).6. The method as claimed in claim 5, wherein, in step (e), the referencesequence number is modified into a value set up in a second field of thecurrently received data.
 7. The method as claimed in claim 5, furthercomprising the steps of: (g) confirming whether or not a currentlyreceived bit stream is transmitted next to a previously received bitstream when it is confirmed in step (b) that a bit stream of currentlyreceived audio data is not a first bit stream of an audio frame; (h)storing the received data when the currently received bit stream istransmitted next to the previously received bit stream; (i) resettingthe reference sequence number; and (j) returning to step (b).
 8. Themethod as claimed in claim 7, wherein, in step (i), the referencesequence number is modified into a value set in a second field of thecurrently received data.
 9. The method as claimed in claim 7, furthercomprising the steps of: (k) deleting the received data when it isconfirmed in step (g) that the currently received bit stream is nottransmitted next to the previously received bit stream; and (l)resetting the reference sequence number and returning to step (b). 10.The method as claimed in claim 9, wherein, in step (1), the referencesequence number is modified into a maximum value being able to be set ina second field.
 11. A method for correcting a codec setup error in amobile communication terminal by analyzing voice slot data received froma mobile communication network while a multimedia audio codec is set upin the mobile communication terminal and identifying a codec used toencode sound source data inserted into a data field of the voice slotdata to confirm whether or not the mobile communication terminal and themobile communication network have an identical codec, the methodcomprising the steps of: (a) checking a sequence number inserted intothe data field of every received voice slot data and determining thatthere is a codec setup error when the voice slot data is not received inorder; and (b) replacing the multimedia audio codec with a voice codec,when it has been determined that there is a codec setup error.
 12. Themethod as claimed in claim 11, wherein step (a) includes the steps of:(a1) designating a first sequence number inserted into the data field ofthe voice slot data as a reference sequence number and designating acodec counter as 0 when the mobile communication terminal receives thevoice slot data from the mobile communication network; (a2) determiningwhether or not a second sequence number of next voice slot data islarger than the reference sequence number by 1 when the next voice slotdata is received; (a3) designating the second sequence number as thereference sequence number when the second sequence number is larger thanthe reference sequence number by 1 and returning to step (a2); (a4)determining whether or not the second sequence number is 0 when thesecond sequence number is not larger than the reference sequence numberby 1, designating the second sequence number as the reference sequencenumber when the second sequence number is 0, and returning to step (a2);(a5) increasing the codec counter by 1 when the second sequence numberis not 0, determining whether or not the codec counter is larger than orequal to a predetermined number N, designating the second sequencenumber as the reference sequence number when the codec counter issmaller than the predetermined number N, and returning to step (a2); and(a6) determining that there is a codec setup error when the codeccounter is larger than or equal to the predetermined number N.
 13. Themethod as claimed in claim 12, wherein the predetermined number N islarger than or equal to
 2. 14. The method as claimed in claim 11,wherein the voice slot data is adapted to be used in a CDMA (CodeDivision Multiple Access) mobile communication network and has a slotperiod of 20 ms, and the data field has a size of 171 bits.
 15. Themethod as claimed in claim 14, wherein the sound source data is mappedonto 168 bits (21 bytes) of the data field, and remaining 3 bits of thedata field are used as a sequence number field, the sequence numberbeing inserted into the sequence number field.
 16. The method as claimedin claim 11, wherein the multimedia audio codec is an HE-AAC codec. 17.The method as claimed in claim 11, wherein the voice codec is an EVRCcodec.
 18. A method for correcting a codec setup error in a mobilecommunication terminal by analyzing voice slot data received from amobile communication network while a voice codec is set up in the mobilecommunication terminal and identifying a codec used to encode soundsource data inserted into a data field of the voice slot data so as toconfirm whether or not the mobile communication terminal and the mobilecommunication network have an identical codec, the method comprising thesteps of: (a) checking a sequence number SEQ inserted into the datafield of every received voice slot data and determining that there is acodec setup error when the voice slot data is received in order; and (b)replacing the voice codec with a multimedia audio codec, when it isdetermined that there is a codec setup error.
 19. The method as claimedin claim 18, wherein step (a) includes the steps of: (a1) designating afirst sequence number inserted into the data field of the voice slotdata as a reference sequence number and designating a codec counter as 0when the mobile communication terminal receives the voice slot data fromthe mobile communication network; (a2) determining whether or not asecond sequence number of next voice slot data is larger than thereference sequence number by 1 when the next voice slot data has beenreceived, increasing the codec counter by 1 when the second sequencenumber is larger than the reference sequence number by 1, and proceedingto step (a4); (a3) determining whether or not the second sequence numberis 0 when the second sequence number is not larger than the referencesequence number by 1, designating the second sequence number as thereference number when the second sequence number is 0, and returning tostep (a2); (a4) determining whether or not the codec counter is largerthan or equal to a predetermined number N, designating the secondsequence number as the reference sequence number when the codec counteris smaller than the predetermined number N, and proceeding to step (a2);and (a5) determining that there is a codec setup error when the codeccounter is larger than or equal to the predetermined number N.
 20. Themethod as claimed in claim 19, wherein the predetermined number N islarger than or equal to
 2. 21. The method as claimed in claim 18,wherein the voice slot data is adapted to be used in a CDMA mobilecommunication network and has a slot period of 20 ms, and the data fieldhas a size of 171 bits.
 22. The method as claimed in claim 21, whereinthe sound soured data is mapped onto 168 bits (21 bytes) of the datafield, and remaining 3 bits of the data field are used as a sequencenumber field, the sequence number being inserted into the sequencenumber field.
 23. The method as claimed in claim 18, wherein themultimedia audio codec is an HE-AAC codec.
 24. The method as claimed inclaim 18, wherein the voice codec is an EVRC codec.
 25. A mobilecommunication terminal for correcting a codec setup error by analyzingvoice slot data received from a mobile communication network andidentifying a codec used to encode sound source data inserted into adata field of the voice slot data to confirm whether or not the mobilecommunication terminal and the mobile communication network have anidentical codec, the mobile communication terminal comprising: a voicecodec for decoding and outputting voice data being inserted into thevoice slot data and being transmitted during voice communication; amultimedia audio codec for decoding and outputting music datatransmitted by a CRBT (Color Ring Back Tone) server providing a CRBTservice, the music data being inserted into the voice slot data andbeing transmitted; and a codec control unit for performing a firstfunction of driving the voice codec when a control message informing thestart of voice communication is received from the mobile communicationnetwork, driving the multimedia audio codec when a control messageinforming the transmission of a ringback tone is received, checking asequence number inserted into the data field of every received voiceslot data when the voice slot data is received while the multimediaaudio codec is in operation, determining that there is a codec setuperror when the voice slot data is not received in order, restricting theoperation of the multimedia audio codec, and letting the voice codec inoperation, the codec control unit performing a second function ofchecking a sequence number inserted into the data field of everyreceived voice slot data when the voice slot data is received while thevoice codec is in operation, determining that there is a codec setuperror when the voice slot data is received in order, restricting theoperation of the voice codec, and driving the multimedia audio codec.26. The mobile communication terminal as claimed in claim 25, whereinthe first function is performed by way of the steps of: (a1) designatinga first sequence number inserted into a data field of voce slot data asa reference sequence number and designating a codec counter as 0 whenthe voice slot data is received from the mobile communication network;(a2) determining whether or not a second sequence number of next voiceslot data is larger than the reference sequence number by 1 when thenext voice slot data is received; (a3) designating the second sequencenumber as the reference sequence number when the second sequence numberis larger than the reference sequence number by 1 and returning to step(a2); (a4) determining whether or not the second sequence number is 0when the second sequence number is not larger than the referencesequence number by 1, designating the second sequence number as thereference sequence number when the second sequence number is 0, andreturning to step (a2); (a5) increasing the codec counter by 1 when thesecond sequence number is not 0, determining whether or not the codeccounter is larger than or equal to a predetermined number N, designatingthe second sequence number as the reference sequence number when thecodec counter is smaller than the predetermined number N, and returningto step (a2); and (a6) determining that there is a codec setup errorwhen the codec counter is larger than or equal to the predeterminednumber N, restricting the operation of the multimedia audio codes anddriving the voice codec.
 27. The mobile communication terminal asclaimed in claim 25, wherein the second function is performed throughthe steps of: (b1) designating a first sequence number inserted into adata field of voice slot data as a reference sequence number anddesignating a codec counter as 0 when the voice slot data is receivedfrom the mobile communication network; (b2) determining whether or not asecond sequence number of next voice slot data is larger than thereference sequence number by 1 when the next voice slot data is receivedincreasing the codec counter by 1 when the second sequence number islarger than the reference sequence number by 1, and proceeding to step(b4); (b3) determining whether or not the second sequence number is 0when the second sequence number is not larger than the referencesequence number by 1, designating the second sequence number as thereference number when the second sequence number is 0, and returning tostep (b2); (b4) determining whether or not the codec counter is largerthan or equal to a predetermined number N, designating the secondsequence number as the reference sequence number when the codec counteris smaller than the predetermined number N, and returning to step (b2);and (b5) determining that there is a codec setup error when the codeccounter is larger than or equal to the predetermined number N,restricting the operation of the voice codec, and driving the multimediaaudio codec.